Why SIP client has sparked interest in the telephony market?
It is an open standard.
Since it is text-based, anyone can read a SIP message that is passed between two endpoints in a SIP session. The protocol is flexible.
It allows users to manage and modulate every call that enters their phone system.
SIP is content-agnostic. It is used for establishing sessions for voice, video, messaging applications, and file transfers.
SIP’s main function is to set up calls and end them.This is a 5-part process
User availability
Tells the protocol whether the party that is being called is available to take it.
Session setup
Connects a call from one user to the other.
User location
Tragofone uses APNS to send push notifications to iOS devices and Google FCM for Android phones.
User capabilities
Determines the media that will be used for the call.
Session management
Running of the call session. For example: transferring and ending a call.
How SIP powers a VoIP session call flow
Registration
When a user agent launches, it has to register with a VoIP SIP server so it can be found by other users. This happens by using the SIP Register request message.
Establishing a call
SIP invite request: The SIP invite initiates an attempt to establish a call. This is sent from the caller to the SIP server where it finds the call receiver’s number. Once the number is located, the invite is forwarded.
SIP response 100: A message is sent from the SIP server to the end user’s number in order to confirm the invite request.
SIP Response 180: This message is a confirmation that the invite has been received and the user agent is alerting the user.
SIP Response 200: A 200 response is sent to confirm a call when it is answered by the end-user. This message results in an exchange and negotiation of VoIP call parameters. SIP Ack Request: The caller confirms the call by sending an Ack request to the number called.
VoIP Call
The VoIP call is transmitted between the user agents using RTP (real-time transport protocol), which is also used for sending audio and video data over IP networks.
Terminating a call
When a user ends a call, a SIP softphone client Bye request is sent. The call can be ended from either side of the line. The termination is confirmed by the other user agent with a SIP 200 status code response.
How SIP adds value to VoIP phone system
Provides crystal clear images in long-distance VoIP video conference calls.
Easy way to send files and documents.
Builds on VoIP’s capabilities by adding video and data transfer potential to it.
Not only does SIP set up the initiation of contact between devices, but it also allows
users to add more parties to a call, or switch between communication methods.
How to get your Business ready for SIP Calling
Robust network infrastructure
Verify that your existing hardware, routers, and wireless network is capable to handle Quality of Service (QoS) settings and supports a SIP-based environment.
Bandwidth & VoIP-compatible
Add higher bandwidth than your current data connection for a smooth transition. Upgrade to VoIP phone system. Some traditional phones can be upgraded with VoIP adapters
SIP-compatible PBX
You may need to upgrade your PBX phone system to make it compatible with SIP calling. It is also advisable to switch to a full cloud-based communications solution to keep the transition smooth.
SIP Benefits for Businesses
Make long-distance calls at the price of local calls.
SIP phone calls can be received on all kinds of mobile and laptop devices.
SIP phone calling makes it easy to route to mobile phones or other workstations.
Your entire suite of communications—internal and external—can be combined into one easy-to-use, computer-based system.
With SIP calling, your business can hire a remote workforce that is not tied to a desk.
Add and manage SIP lines seamlessly through a dedicated web-based portal.
There are several tools available to monitor and resolve issues in quick time.
SIP softphone can give you significant returns on a relatively small investment.